An in-room acoustic magnitude response of an audio system, in the absence of corrective or compensatory measures, may be dominated by resonant effects. Low-frequency “standing waves,” whose resonant frequencies are generally affected by the major dimensions and shape of an acoustic space, may impose severe peaks (e.g., pressure maxima, known as “antinodes”) and notches (e.g., pressure minima or “nodes”) on an in-room acoustic response of a loudspeaker system within the acoustic space. The mechanisms by which such resonant modes are excited are fairly well understood, and many different methods may be employed to control these resonant modes so as to achieve a smooth measured in-room acoustic magnitude response that is free of peaks and notches. Such smooth, flat acoustic response curves, indicative of an audio system that neither favors nor fails to reproduce any portions of its passband, are typically preferred over a magnitude response characterized by severe peaks and notches. However, the transient response of the system, i.e., the ability of the system to respond quickly and accurately as the input signal starts and stops, is also an important part of perceived system performance. This aspect of system performance is both difficult to measure and very difficult to preserve when applying known techniques for the control of room induced resonant modes.
Conventional methods for achieving a smooth in-room acoustic magnitude response have focused on various techniques for measuring the uncorrected in-room response and the application of magnitude response equalization using various devices. Many methods have been proposed for correction of the in-room magnitude response of an audio reproduction system. Typically, these methods include a device to measure the in-room magnitude response of the audio system at a specific listening location, a device to derive a corrective filter, and a device to apply a corrective filter to an input signal prior to reproduction by the audio system. The corrective filter is typically composed of narrow band attenuation filters located at the frequencies where the resonant behavior of the room causes peaks in the in-room magnitude response. Appropriate attenuation of these peaks will produce a more or less smooth and flat in-room magnitude response. There are, however, two significant shortcomings of these systems. First, the modal density of the room, particularly at mid and higher frequencies, may be so high as to make the corrective filters extremely complex and impractical to implement even with digital techniques. Second, all resonant modes take a certain amount of time to establish themselves in the room. This means that the initial sound, immediately following the onset of the input signal, is unaffected by the resonant behavior of the room. However, in some systems the band attenuation filters of correction methods are applied continuously to the input signals, and attenuate both the initial and steady state portions of the sound. This leads to poor transient response and the subjective impression that the system does not exhibit dynamics or impact.
Some prior methods have recognized the need to accommodate both time and frequency domains room correction methods so as to preserve proper transient response. For example, some employ two corrective filters with different characteristics. One corrective filter modifies the initial sound and the other corrective filter, after a delay, modifies the later or steady state sound. This may help to preserve proper transient response of the initial sound. However, the method requires derivation of two possibly complex correction filters and manual adjustment of the delay for application of the second filter. Another prior method digitally creates a corrective filter in the time domain using Frequency Induced Resonance (FIR) type filters. In theory, such a digital system for time domain modification of the input signal should provide perfect correction of the in-room response of the system in both the time and frequency domains. However, subsequent advances in digital signal processing theory have shown that the characteristics of FIR filters make implementation of this method impractical even with current digital signal processing (DSP) hardware. In addition, as is now well known, digital time domain techniques are limited in their ability to correct signal problems resulting from stored or delayed energy, such as in room resonances.
Therefore, what is needed is a system and method through which an in-room performance of an audio system, as measured in part by an associated acoustic in-room magnitude response, may be improved, without compromising dynamic or transient performance.